I am using Asterisk & java application for voip project in centos.
java app use for keep Database with help of mysql & asterisk & monitor everything…
Now i am trying to use open source soft-phone on same machine where asterisk present.
I try pjsua but when ever asterisk on pjsua not work & when ever asterisk stop pjsua works…
Now i am also trying to find out in Asterisk there is App-Konference…is there way to fetch audio coming from 1 sip phone to asterisk & we can send it. like softphone do?
Means person who handle asterisk server he also talk with other sip phones… by headphone which attach to server’s sound card…
or by vbox i can keep running one more centos in same server where soft phone install & in main machine Asterisk, mysql,java etc running…& person can talk with same physical machine to sip phone…
Is this few possibilities i thing…but which one really possible not know?
Any Idea or clue can i get from here?
now as per david55 says
I want to ask…i have 1 pc which i am using as server of Asterisk & same containing mysql, java program & few sip phones i have…now when i establish all things as per various forums instructions…i can do different calls from sip phones via asterisk.
When i use 1 laptop which holds softphone, i can manage sip phone do calls softphone & vice versa…
Now i am trying Softphone & Asterisk etc on same pc. So can sip phone do call server’s softphone?
Following i understood after doing R and D.
Problem is if server ip 10.111.17.4 then for all sip phone asterisk’s ip is same & asterisk redirect calls like 10.111.17.10 calling 10.111.17.20 then 10.111.17.4 get request from 10.111.17.10 & it redirect it to 10.111.17.20…now how asterisk redirect self ip…something like that i understood. Is it right?
Now mean time I read about oss.conf via google but not properly understood…!
Can i get some clue or help or idea?